openalbridge/loaders.c
changeset 2191 20c62f787a4d
child 2200 8192be6e3aef
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/openalbridge/loaders.c	Wed Jun 24 15:59:32 2009 +0000
@@ -0,0 +1,212 @@
+/*
+ * OpenAL Bridge - a simple portable library for OpenAL interface
+ * Copyright (c) 2009 Vittorio Giovara <vittorio.giovara@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <stdint.h>
+#include "al.h"
+#include "alc.h"
+#include "loaders.h"
+#include "endianness.h"
+#include "wrappers.h"
+
+#ifdef __CPLUSPLUS
+extern "C" {
+#endif 
+	
+	extern int ov_open(FILE *f,OggVorbis_File *vf,char *initial,long ibytes);
+	extern long ov_read(OggVorbis_File *vf,char *buffer,int length,int bigendianp,int word,int sgned,int *bitstream);
+	extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i);
+	extern long ov_read(OggVorbis_File *vf,char *buffer,int length,int bigendianp,int word,int sgned,int *bitstream);
+	extern vorbis_info *ov_info(OggVorbis_File *vf,int link);
+	extern vorbis_comment *ov_comment(OggVorbis_File *f, int num);
+	
+	int load_WavPcm (const char *filename, ALenum *format, uint8_t** data, ALsizei *bitsize, ALsizei *freq) {
+		WAV_header_t WAVHeader;
+		FILE *wavfile;
+		int t, n = 0;
+		
+		wavfile = Fopen(filename, "rb");
+		
+		fread(&WAVHeader.ChunkID, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.ChunkSize, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.Format, sizeof(uint32_t), 1, wavfile);
+		
+#ifdef DEBUG
+		fprintf(stderr, "ChunkID: %X\n", invert_endianness(WAVHeader.ChunkID));
+		fprintf(stderr, "ChunkSize: %d\n", WAVHeader.ChunkSize);
+		fprintf(stderr, "Format: %X\n", invert_endianness(WAVHeader.Format));
+#endif
+		
+		fread(&WAVHeader.Subchunk1ID, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.Subchunk1Size, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.AudioFormat, sizeof(uint16_t), 1, wavfile);
+		fread(&WAVHeader.NumChannels, sizeof(uint16_t), 1, wavfile);
+		fread(&WAVHeader.SampleRate, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.ByteRate, sizeof(uint32_t), 1, wavfile);
+		fread(&WAVHeader.BlockAlign, sizeof(uint16_t), 1, wavfile);
+		fread(&WAVHeader.BitsPerSample, sizeof(uint16_t), 1, wavfile);
+		
+#ifdef DEBUG
+		fprintf(stderr, "Subchunk1ID: %X\n", invert_endianness(WAVHeader.Subchunk1ID));
+		fprintf(stderr, "Subchunk1Size: %d\n", WAVHeader.Subchunk1Size);
+		fprintf(stderr, "AudioFormat: %d\n", WAVHeader.AudioFormat);
+		fprintf(stderr, "NumChannels: %d\n", WAVHeader.NumChannels);
+		fprintf(stderr, "SampleRate: %d\n", WAVHeader.SampleRate);
+		fprintf(stderr, "ByteRate: %d\n", WAVHeader.ByteRate);
+		fprintf(stderr, "BlockAlign: %d\n", WAVHeader.BlockAlign);
+		fprintf(stderr, "BitsPerSample: %d\n", WAVHeader.BitsPerSample);
+#endif
+		
+		do { //remove useless header chunks (plenty room for improvements)
+			t = fread(&WAVHeader.Subchunk2ID, sizeof(uint32_t), 1, wavfile);
+			if (invert_endianness(WAVHeader.Subchunk2ID) == 0x64617461)
+				break;
+			if (t <= 0) { //eof found
+				fprintf(stderr, "ERROR: wrong WAV header\n");
+				return AL_FALSE;
+			}
+		} while (1);
+		fread(&WAVHeader.Subchunk2Size, sizeof(uint32_t), 1, wavfile);
+		
+#ifdef DEBUG
+		fprintf(stderr, "Subchunk2ID: %X\n", invert_endianness(WAVHeader.Subchunk2ID));
+		fprintf(stderr, "Subchunk2Size: %d\n", WAVHeader.Subchunk2Size);
+#endif
+		
+		*data = (uint8_t*) malloc (sizeof(uint8_t) * WAVHeader.Subchunk2Size);
+		
+		//this could be improved
+		do {
+			n += fread(&((*data)[n]), sizeof(uint8_t), 1, wavfile);
+		} while (n < WAVHeader.Subchunk2Size);
+		
+		fclose(wavfile);	
+		
+#ifdef DEBUG
+		fprintf(stderr, "Last two bytes of data: %X%X\n", (*data)[n-2], (*data)[n-1]);
+#endif
+		
+		/*remaining parameters*/
+		//Valid formats are AL_FORMAT_MONO8, AL_FORMAT_MONO16, AL_FORMAT_STEREO8, and AL_FORMAT_STEREO16. 
+		if (WAVHeader.NumChannels == 1) {
+			if (WAVHeader.BitsPerSample == 8)
+				*format = AL_FORMAT_MONO8;
+			else {
+				if (WAVHeader.BitsPerSample == 16)
+					*format = AL_FORMAT_MONO16;
+				else {
+					fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
+					return AL_FALSE;
+				}
+			} 
+		} else {
+			if (WAVHeader.NumChannels == 2) {
+				if (WAVHeader.BitsPerSample == 8)
+					*format = AL_FORMAT_STEREO8;
+				else {
+					if (WAVHeader.BitsPerSample == 16)
+						*format = AL_FORMAT_STEREO16;
+					else {
+						fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
+						return AL_FALSE;
+					}				
+				}
+			} else {
+				fprintf(stderr, "ERROR: wrong WAV header - format value\n");
+				return AL_FALSE;
+			}
+		}
+		
+		*bitsize = WAVHeader.Subchunk2Size;
+		*freq = WAVHeader.SampleRate;
+		return AL_TRUE;
+	}
+	
+	int load_OggVorbis (const char *filename, ALenum *format, uint8_t**data, ALsizei *bitsize, ALsizei *freq) {
+		//implementation inspired from http://www.devmaster.net/forums/showthread.php?t=1153
+		FILE			*oggFile;		// ogg handle
+		OggVorbis_File  oggStream;		// stream handle
+		vorbis_info		*vorbisInfo;	// some formatting data
+		vorbis_comment	*vorbisComment;	// other less useful data
+		int64_t			pcm_length;		// length of the decoded data
+		int size = 0;
+		int section, result, i;
+		
+		oggFile = Fopen(filename, "rb");
+		result = ov_open(oggFile, &oggStream, NULL, 0);
+		//TODO: check returning value of result
+		
+		vorbisInfo = ov_info(&oggStream, -1);
+		pcm_length = ov_pcm_total(&oggStream,-1) << vorbisInfo->channels;	
+		
+#ifdef DEBUG
+		vorbisComment = ov_comment(&oggStream, -1);
+		fprintf(stderr, "Version: %d\n", vorbisInfo->version);
+		fprintf(stderr, "Channels: %d\n", vorbisInfo->channels);
+		fprintf(stderr, "Rate (Hz): %d\n", vorbisInfo->rate);
+		fprintf(stderr, "Bitrate Upper: %d\n", vorbisInfo->bitrate_upper);
+		fprintf(stderr, "Bitrate Nominal: %d\n", vorbisInfo->bitrate_nominal);
+		fprintf(stderr, "Bitrate Lower: %d\n", vorbisInfo->bitrate_lower);
+		fprintf(stderr, "Bitrate Windows: %d\n", vorbisInfo->bitrate_window);
+		fprintf(stderr, "Vendor: %s\n", vorbisComment->vendor);
+		fprintf(stderr, "PCM data size: %d\n", pcm_length);
+		fprintf(stderr, "# comment: %d\n", vorbisComment->comments);
+		for (i = 0; i < vorbisComment->comments; i++)
+			fprintf(stderr, "\tComment %d: %s\n", i, vorbisComment->user_comments[i]);
+#endif
+		
+		//allocates enough room for the decoded data
+		*data = (uint8_t*) malloc (sizeof(uint8_t) * pcm_length);
+		
+		//there *should* not be ogg at 8 bits
+		if (vorbisInfo->channels == 1)
+			*format = AL_FORMAT_MONO16;
+		else {
+			if (vorbisInfo->channels == 2)
+				*format = AL_FORMAT_STEREO16;
+			else {
+				fprintf(stderr, "ERROR: wrong OGG header - channel value (%d)\n", vorbisInfo->channels);
+				return AL_FALSE;
+			}
+		}
+		
+		while(size < pcm_length)	{
+			//ov_read decodes the ogg stream and storse the pcm in data 
+			result = ov_read (&oggStream, *data + size, pcm_length - size, 0, 2, 1, &section);
+			if(result > 0) {
+				size += result;
+			} else {
+				if (result == 0)
+					break;
+				else { 
+					fprintf(stderr, "ERROR: end of file from OGG stream\n");
+					return AL_FALSE;
+				}
+			}
+		}
+		
+		//records the last fields
+		*bitsize = size;
+		*freq = vorbisInfo->rate;
+		return AL_TRUE;
+	}
+	
+#ifdef __CPLUSPLUS
+}
+#endif